I'm not an expert in the field of sound recording, and I need Your
help in one of the projects I'm currently working at.

I need to be able to
record sound sequences - each one 40 to 60 seconds long, at very high speed. The sound is
beyond the human hearing range on both ends - from 3 Hz to 55 kHz, and everything in
between. But this is not an issue.

By "speed" I mean sampling rate, probably
(correct me if I'm wrong). What I have now is a very simple equipment (well, comparing to
pro tech), that allows me to record at 96 kHz sampling rate and 16-bit depth.

Every sequence need to be slowed down and played at 44.1 kHz and no less. I'm looking
for a way to record my input and then slow it down without any loss of quality and without
any significant software "intrusion" into the signal. The problem is, right now I can only
slow it down to about half of the original speed before there is no more data in the
signal. And I need to be able to slow it down much more - by much more I mean "as much as
humanely possible without having to be a billionaire". The more the better. Anything
better than half, meaning twice the playing time of the original signal is good. 50x
slower or more would be ideal.

So I want to ask You:

- is this
possible?- if yes, how expensive would it be?- what kind of equipment would I
need?- what is the current technological limit for samplig rate of an audio
waveform, and therefore for lossless slowing down of the signal?

If yu have a PC then a cheap DAW program such as Reaper will gve you most of the tools you
need. As Jumpy says, you really want a 192kHz recording rate, then you can play that back
at 44.1kHz to give you a 4.3:1 ratio.This should bring the 55kHz sounds down to 12.6kHz -
within the audible range.

If you want to get the frequency even lower you can
play back the 192kHz file at slower rates than 44.1 if the DAW sofware and your audio
interface allow it (e.g. 22.05kHz). Then you can take the resulting output file and then
convert the file into a 44.1kHz format (this is not the same as simply playing it back at
44.1kHz).

Beyond this, you are looking at pitch shifting algorithms which may
give you another octave downward shift of reasonable sound quality.

But all
this down shifting will put an awful lot of sound energy in the low and very low frequency
range. You will easily blow speakers if you are not very careful and any resulting files
should go through a high pass filter that cuts out sound below the frequency range of your
playback speakers.

Likewise if you record very low frequencies and speed up the
playback to get them into an audiable frequency range, you can shift a lot of sound energy
into very high frequencies which won't do the HF components of yur playback system much
good either, so a low pass filter is required here.

And as Jumpy again said,
getting a single suitable mic to cover that range could be a very tall order indeed.